Index: webrtc/audio/audio_send_stream.h |
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
index 065fee78947e16b70f8ce9403a0c459a23e76de7..881c68ed4db55d49987c8d064b7d9f8bd1162207 100644 |
--- a/webrtc/audio/audio_send_stream.h |
+++ b/webrtc/audio/audio_send_stream.h |
@@ -19,7 +19,7 @@ |
#include "webrtc/call/audio_send_stream.h" |
#include "webrtc/call/audio_state.h" |
#include "webrtc/call/bitrate_allocator.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
#include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" |
namespace webrtc { |
@@ -44,7 +44,8 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
RtpTransportControllerSendInterface* transport, |
BitrateAllocator* bitrate_allocator, |
RtcEventLog* event_log, |
- RtcpRttStats* rtcp_rtt_stats); |
+ RtcpRttStats* rtcp_rtt_stats, |
+ const rtc::Optional<RtpState>& suspended_rtp_state); |
~AudioSendStream() override; |
// webrtc::AudioSendStream implementation. |
@@ -74,6 +75,8 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
const webrtc::AudioSendStream::Config& config() const; |
void SetTransportOverhead(int transport_overhead_per_packet); |
+ RtpState GetRtpState() const; |
+ |
private: |
VoiceEngine* voice_engine() const; |
@@ -111,6 +114,9 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
TransportFeedbackPacketLossTracker packet_loss_tracker_ |
GUARDED_BY(&packet_loss_tracker_cs_); |
+ RtpRtcp* rtp_rtcp_module_; |
+ rtc::Optional<RtpState> const suspended_rtp_state_; |
+ |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
}; |
} // namespace internal |